There exist different mechanisms to convey speech between operators. The transport can be a CS—(Circuit Switched), a TDM—(Time Division Multiplexing) or an IP-based (Internet Protocol) transport. If the IP-based transport is used, the protocol can be a UDP (User Datagram Protocol), a TCP (Transmission control Protocol) protocol or an SCTP (Stream Control Transmission Protocol) protocol. For transporting video data, there exist even more possibilities. This means that in many signal traffic situations between operators at least one of the end points of a communication, or even both of the end points must carry out conversions to make the connection and the transport of a signal possible. Every conversion requires a network or terminal support, increases delays, makes a quality of the signal worse etc. Thus in an ideal situation the number of the conversions is as low as possible. Traditionally, this has been solved in such a way that the operators themselves have carried out the required conversions in their own networks.
So far this problem has not played a major role, because a traditional speech transmission has in practice been the only widely used service and the number of conversions has been quite low due to the fact that nearly all operators use TDM connections.
One of the disadvantages associated with the prior art arrangement is that when different signals and data will be used in larger scale, the traditional model will become difficult to use, because different operators use different models for conversions and thus many different options for conversions must be planned, realized, supported and stored. Furthermore, there may be no clear common agreed model to indicate which end, end point or which operator makes the conversion(s). If a first end changes its system, also a second end must update its configuration(s).